[asterisk-users] retransmision error con asterisk 1.4.24.1

Martin asterisklist at callthem.info
Sun Apr 12 23:07:51 CDT 2009 

1) your asterisk box talks to OpenSIPS
2) in that case OpenSIPS should traverse NAT
3) you should not do nat=yes for that device since Asterisk talks to
OpenSIPS (but then it might not matter)

Either take OpenSIPS out of the way or configure NAT traversal w/media
and it should work

Martin

On Sun, Apr 12, 2009 at 9:22 PM, troxlinux <xserverlinux at gmail.com> wrote:
> uff  , no me fije que envié un mensaje en español a la lista de ingles ...
>
> I send sip log
>
>
> ---
> Retransmitting #2 (NAT) to 192.168.10.3:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
> Via: SIP/2.0/UDP
> 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
> Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
> From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
> To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
> Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
> CSeq: 30032 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:*981 at 192.168.10.3:5070>
> Content-Type: application/sdp
> Content-Length: 258
>
> v=0
> o=root 3005 3005 IN IP4 192.168.10.3
> s=session
> c=IN IP4 192.168.10.3
> t=0 0
> m=audio 13584 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
> Retransmitting #3 (NAT) to 192.168.10.3:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
> Via: SIP/2.0/UDP
> 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
> Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
> From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
> To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
> Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
> CSeq: 30032 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:*981 at 192.168.10.3:5070>
> Content-Type: application/sdp
> Content-Length: 258
>
> v=0
> o=root 3005 3005 IN IP4 192.168.10.3
> s=session
> c=IN IP4 192.168.10.3
> t=0 0
> m=audio 13584 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
>    -- <SIP/111-08d20da8> Playing 'vm-password' (language 'es')
> Retransmitting #4 (NAT) to 192.168.10.3:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
> Via: SIP/2.0/UDP
> 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
> Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
> From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
> To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
> Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
> CSeq: 30032 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:*981 at 192.168.10.3:5070>
> Content-Type: application/sdp
> Content-Length: 258
>
> v=0
> o=root 3005 3005 IN IP4 192.168.10.3
> s=session
> c=IN IP4 192.168.10.3
> t=0 0
> m=audio 13584 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
>    -- <SIP/111-08d20da8> Playing 'vm-youhave' (language 'es')
> Reliably Transmitting (NAT) to 192.168.10.3:5060:
> OPTIONS sip:192.168.10.3 SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport
> From: "asterisk" <sip:asterisk at 192.168.10.3:5070>;tag=as690b573d
> To: <sip:192.168.10.3>
> Contact: <sip:asterisk at 192.168.10.3:5070>
> Call-ID: 36b27df370b95454445ab61d5c8c251b at 192.168.10.3
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Date: Mon, 13 Apr 2009 02:20:27 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
>
> ---
> twoxserver*CLI>
> <--- SIP read from 192.168.10.3:5060 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 192.168.10.3:5070;branch=z9hG4bK63ca3652;rport=5070
> From: "asterisk" <sip:asterisk at 192.168.10.3:5070>;tag=as690b573d
> To: <sip:192.168.10.3>;tag=d4e9e39d125187795ad79ae40f9b4f9f.478d
> Call-ID: 36b27df370b95454445ab61d5c8c251b at 192.168.10.3
> CSeq: 102 OPTIONS
> Server: OpenSIPS (1.5.0-notls (i386/linux))
> Content-Length: 0
>
>
> <------------->
> --- (8 headers 0 lines) ---
> Really destroying SIP dialog
> '36b27df370b95454445ab61d5c8c251b at 192.168.10.3' Method: OPTIONS
>    -- <SIP/111-08d20da8> Playing 'digits/6' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'vm-messages' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'vm-first' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'vm-message' (language 'es')
>  == Parsing '/var/spool/asterisk/voicemail/default/111/INBOX/msg0000.txt':
> Found
> Retransmitting #5 (NAT) to 192.168.10.3:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
> Via: SIP/2.0/UDP
> 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
> Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
> From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
> To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
> Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
> CSeq: 30032 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:*981 at 192.168.10.3:5070>
> Content-Type: application/sdp
> Content-Length: 258
>
> v=0
> o=root 3005 3005 IN IP4 192.168.10.3
> s=session
> c=IN IP4 192.168.10.3
> t=0 0
> m=audio 13584 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
>    -- <SIP/111-08d20da8> Playing 'vm-received' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/yesterday' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/at' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/8' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/30' (language 'es')
> Retransmitting #6 (NAT) to 192.168.10.3:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.10.3;branch=z9hG4bKf4e2.2a85e302.0;received=192.168.10.3
> Via: SIP/2.0/UDP
> 192.168.10.23:5060;rport=5060;received=192.168.10.23;branch=z9hG4bKd2f59bd22938299d
> Record-Route: <sip:192.168.10.3;lr=on;ftag=a363e3a864940c0e>
> From: "Lucy" <sip:111 at 192.168.10.3>;tag=a363e3a864940c0e
> To: <sip:*981 at 192.168.10.3>;tag=as3a76126d
> Call-ID: 5c4fa1fdece6f915 at 192.168.10.23
> CSeq: 30032 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Contact: <sip:*981 at 192.168.10.3:5070>
> Content-Type: application/sdp
> Content-Length: 258
>
> v=0
> o=root 3005 3005 IN IP4 192.168.10.3
> s=session
> c=IN IP4 192.168.10.3
> t=0 0
> m=audio 13584 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> ---
>    -- <SIP/111-08d20da8> Playing 'digits/and' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/9' (language 'es')
>    -- <SIP/111-08d20da8> Playing 'digits/p-m' (language 'es')
>    -- <SIP/111-08d20da8> Playing
> '/var/spool/asterisk/voicemail/default/111/INBOX/msg0000' (language
> 'es')
> [Apr 12 20:20:36] WARNING[3528]: app_voicemail.c:5619 play_message:
> Playback of message
> /var/spool/asterisk/voicemail/default/111/INBOX/msg0000 failed
>    -- <SIP/111-08d20da8> Playing 'vm-advopts' (language 'es')
> [Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1976 retrans_pkt: Maximum
> retries exceeded on transmission 5c4fa1fdece6f915 at 192.168.10.23 for
> seqno 30032 (Critical Response) -- See doc/sip-retransmit.txt.
> [Apr 12 20:20:38] WARNING[3140]: chan_sip.c:1998 retrans_pkt: Hanging
> up call 5c4fa1fdece6f915 at 192.168.10.23 - no reply to our critical
> packet (see doc/sip-retransmit.txt).
>  == Spawn extension (netsoluciones, *981, 2) exited non-zero on
> 'SIP/111-08d20da8'
> Really destroying SIP dialog '5c4fa1fdece6f915 at 192.168.10.23' Method: INVITE
>
>
> 2009/4/12 Alex Balashov <abalashov at evaristesys.com>:
>> Mejor que obtengamos un packet capture para investigarlo mas.
>
> sera un bug o algo por el estilo
>
> saludoss
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
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